search for: putland

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. >> >> So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do?

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

..., 2) exited non-zero on 'Local/6001@default-2' show locals <unowned> -- 6001@default U?WVS?|?E -- @s 2 Segmentation fault (core dumped) This segfault Produced same traceback as above The above results were produced by extenstion.conf exten => 1998,1,Macro(stdexten,1234,IAX/kputland) exten => 1999,1,Macro(stdexten,1234,IAX/peracles) exten => 6001,1,Answer exten => 6001,2,Queue(dialer) Executing this command on the manager port. Action: Originate Exten: 1999 Channel: Local/6001 Then running show locals over and over and over again rather quickly. -- Karl Putl...

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

...ices w/ ethernet and fxo/fxs configurations. I found this while out and about and thought it might be interesting. Configurable fxo/fxs with dual ethernet. http://www.tekdigitel.com/website/htmlPages/content/products/product_introductions/introduction_to_V-SERVER_iGATE_Dual_Ethernet.htm -- Karl Putland <karl at putland.linux-site.net>

...[13326]: File chan_zap.c, Line 1746 (zt_answer): Took Zap/1-1 off hook -- Executing SetCallerID("Zap/1-1", "H???j;@,@hf@?? ??? <>") in new stack -- Karl Putland <karl@putland.linux-site.net>

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

Bill I have adtran750 with Vista/Aastra 350, 390, & 480's never seen that one tim -----Original Message----- From: asterisk at billheckel.com <asterisk at billheckel.com> To: asterisk-users at lists.digium.com <asterisk-users at lists.digium.com> Date: February 21, 2003 12:52 PM Subject: [Asterisk-Users] Fun new problem with Aastra 390 ADSI phones >I seem to have

...e forward a detailed description of the issue to me to verify if it is our problem or a dlink problem. I've got a support contact at dlink that is very helpful. If I can verify that it's a dlink issue I'll forward the information about the issue and your contact info to him. -- Karl Putland <karl@putland.linux-site.net>

We want to test capacity of our MeetMe room. The thing that is distinct about this is that the incoming line is being delivered IAX2 to our server across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test both the VoIP Gateway and the MeetMe room performance. You can reach our MeetMe room directly at 1-301-561-9229 If you want to test with us we're thinking maybe 9pm

Hi, I havn't actually tried this yet, but would it be possible to use asterisk as a mgcp <-> h.323 translator? For example, I have mgcp service from Next Gen telephone company. But i only have a h.323 phone. Would there be a way to the mgcp signalling to hit asterisk, and then have it fire the call out h.323? And vice versa? Just brainstorming. Sean Watkins

Just a quick query to find out if there are any other Asterisk users in Colorado. If you're out there, drop me a line off-list. I'd like to start a user group if there is anyone else out there. --Karl -- Karl Putland <karl@putland.linux-site.net>

...isk box is now completely locked up and the phones continue to ring regardless of whether they are answered of hungup. Same scenario with mec2 and all is well with the world. Looks like I won't be using mec3 anytime soon. Is anyone else able to confirm this issue with mec3? --Karl -- Karl Putland <karl@putland.linux-site.net>

...avs/123456_1_1_0.745781945801 RESULT_LINE: 200 result=0 endpos=0 == Spawn extension (activity-alerts, s, 2) exited non-zero on 'IAX2[NuFone@Outgoing1]/2' At the point where it tries to playback the file the warnings are spit out and the call is disconnected. Any ideas? --Karl -- Karl Putland <karl@putland.linux-site.net>

Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? Ta SJ

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

Hi, I'm a new asterisk user developing an AGI application. As part of my application I'd like to play sounds on the server's speakers, but it seems that I can't do this while asterisk is running. When I try to play sounds using the play or aplay command, it blocks until I stop asterisk. My guess is that asterisk is using the sound device and this means that other programs

I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail: